Although most people are not acquainted with dial plans by name, they
have become accustomed to using them. The North American telephone network is
designed around a 10-digit dial plan that consists of area codes and 7-digit
telephone numbers. For telephone numbers located within an area code, a 7-digit
dial plan is used for the public switched telephone network (PSTN). Features
within a telephone switching machine (such as Centrex) allow for the use of a
custom 5-digit dial plan for specific customers who subscribe to that service.
Private branch exchanges (PBXs) also allow for variable length dial plans that
contain three to eleven digits. Dial plans contain specific dialing patterns
for a user who wants to reach a particular telephone number. Access codes, area
codes, specialized codes, and combinations of the numbers of digits dialed are
all a part of any particular dial plan.

Dial plans require knowledge of the customer's network topology,
current telephone number dialing patterns, proposed router/gateway locations,
and traffic routing requirements. If the dial plans are for a private internal
voice network that is not accessed by the outside voice network, the telephone
numbers can be any number of digits.

The dial plan design process begins with the collection of specific
information about the equipment to be installed and the network to which it is
to be connected. Complete a Site Preparation
Checklist for each unit in the network. This information, coupled with a
network diagram, is the basis for the number plan design and corresponding
configurations.

Dial plans are associated with the telephone networks to which they are
connected. They are usually based on numbering
plans and the traffic in terms of the number of voice calls the network
is expected to carry.

For more information about Cisco IOS® dial peers, refer to these
documents:

The North American Numbering Plan (NANP) consists of a 10-digit dial
plan. This is divided into two basic parts. The first three digits refer to the
Numbering Plan Area (NPA), commonly referred to as the "area code." The
remaining seven digits are also divided into two parts. The first three numbers
represent the central office (CO) code. The
remaining four digits represent a station number.

The NPA, or area codes, are provided in this format:

N 0/1/2/3

N is a value of two through nine.

The second digit is a value of zero through eight.

The third digit is a value of zero through nine.

The second digit, when set to a value of zero through eight, is used to
immediately distinguish between 10- and 7-digit numbers. When the second and
third digits are both "one", this indicates a special action.

211 = Reserved.

311 = Reserved.

411 = Directory assistance.

511 = Reserved.

611 = Repair service.

711 = Reserved.

811 = Business office.

911 = Emergency.

Additionally, the NPA codes also support Service Access Codes (SAC).
These codes support 700, 800, and 900 services.

In the early 1960s, the Consultative Committee for International
Telegraph and Telephone (CCITT) developed a numbering plan that divided the
world into nine zones:

1 = North America

2 = Africa

3 = Europe

4 = Europe.

5 = Central and South America

6 = South Pacific

7 = USSR

8 = Far East

9 = Middle East and Southeast Asia

Additionally, each country is assigned a country code (CC) . This is either one, two, or
three digits long. It begins with a zone digit.

The method recommended by the International Telecommunication Union
Telecommunication Standardization Sector (ITU-T) (formerly the CCITT) is set
forth in Recommendation E.123. International format numbers use the plus sign
(+), followed by the country code, then the Subscriber Trunk Dialing (STD)
code, if any (without common STD/area code prefix digits or long distance
access digits), then the local number. These numbers (given as examples only)
describe some of the formats used:

City

Domestic Number

International Format

Toronto, Canada

(416) 872-2372

+ 1 416 872 2372

Paris, France

01 33 33 33 33

+ 33 1 33 33 33 33

Birmingham, UK

(0121) 123 4567

+ 44 121 123 4567

Colon, Panama

441-2345

+ 507 441 2345

Tokyo, Japan

(03) 4567 8901

+ 81 3 4567 8901

Hong Kong

2345 6789

+ 852 2345 6789

In most cases, the initial 0 of an STD code does not form part of the
international format number. Some countries use a common prefix of 9 (such as
Colombia, and formerly Finland). Some countries' STD codes are used as they
are, where prefix digits are not part of the area code (as is the case in North
America, Mexico, and several other countries).

As indicated in the example table, country code "1" is used for the
United States, Canada, and many Caribbean nations under the NANP. This fact is
not as well publicized by American and Canadian telephone companies as it is in
other countries. "1" is dialed first in domestic long distance calls. It is a
coincidence that this is identical to country code 1.

The digits that follow the + sign represent the number as it is dialed
on an international call (that is, the telephone company's overseas dialing
code followed by the international number after the + sign).

The access codes for international dialing depend on the country from
which an international call is placed. The most common international prefix is
00 (followed by the international format number). An ITU-T recommendation
specifies 00 as the preferred code. In particular, the European Union (EU)
nations are adopting 00 as the standard international access code.

Traffic Engineering, as it applies to traditional voice networks,
determines the number of trunks necessary to carry a required amount of voice
calls during a period of time. For designers of a voice over X network, the
goal is to properly size the number of trunks and provision the appropriate
amount of bandwidth necessary to carry the amount of trunks determined.

There are two different types of connections to be aware of. They are
lines and trunks. Lines allow telephone sets to be connected to telephone
switches, like PBXs and CO switches. Trunks connect switches together. An
example of a trunk is a tie line interconnecting PBXs (ignore the use of "line"
in the tie line statement. It is actually a trunk).

Companies use switches to act as concentrators because the number of
telephone sets required are usually greater than the number of simultaneous
calls that need to be made. For example, a company has 600 telephone sets
connected to a PBX. However, it has only fifteen trunks that connect the PBX to
the CO switch.

Traffic Engineering a voice over X network is a five step
process.

The steps are:

Collect the existing voice traffic data.

Categorize the traffic by groups.

Determine the number of physical trunks required to meet the
traffic.

Determine the proper mix of trunks.

Convert the number of erlangs of traffic to packets or cells per
second.

Collect the existing voice traffic.

From the carrier, gather this information:

Peg counts for calls offered, calls abandoned, and all trunks
busy.

Grade of Service (GoS) rating for trunk groups.

Total traffic carried per trunk group.

Phone bills to see the carrier's rates.

The terms used here are covered in more detail in the next few
sections of this document. For best results, get two weeks' worth of traffic.

The internal telecommunications department provides call detail
records (CDR) for PBXs. This information records calls that are offered.
However, it does not provide information on calls that are blocked because all
trunks are busy.

Categorize the traffic by groups.

In most large businesses, it is more cost effective to apply
traffic engineering to groups of trunks that serve a common purpose. For
example, separate inbound customer service calls into a separate trunk group
distinctly different from general outgoing calls.

Start by separating the traffic into inbound and outbound
directions. As an example, group outbound traffic into distances called local,
local long distance, intra-state, inter-state, and so on. It is important to
break the traffic by distance because most tariffs are distance sensitive. For
example, wide-area telephone service (WATS) is a type of service option in the
United States that uses distance bands for billing purposes. Band one covers
adjacent states. It has a lower cost than, for example, a band five service
that encompasses the entire continental United States.

Determine the purpose of the calls. For example, what were the
calls for? Were they used for fax, modem, call center, 800 for customer
service, 800 for voice mail, telecommuters, and so on.

Determine the number of physical trunks required to meet the
traffic needs.

If you know the amount of traffic generated and the GoS required,
calculate the number of trunks required to meet your needs. Use this equation
to calculate traffic flow:

A = C x T

A is the traffic flow. C is the
number of calls that originate during a period of one hour. T
is the average holding time of a call.

C is the number of calls originated, not carried. The information
received from the carrier or from the company's internal CDRs are in terms of
carried traffic and not offered traffic, as is usually provided by PBXs.

The holding time of a call (T) must account for the average time a
trunk is occupied. It must factor in variables other than the length of a
conversation. This includes the time required for dialing and ringing (call
establishment), time to terminate the call, and a method of amortizing busy
signals and non-completed calls. Adding ten percent to sixteen
percent to the length of an average call helps account for these miscellaneous
segments of time.

Hold times based on call billing records might need to be adjusted
based on the increment of billing. Billing records based on one minute
increments overstate calls by 30 seconds on average. For example, a bill that
shows 404 calls totaling 1834 minutes of traffic needs to be adjusted like
this:

In order to provide a "decent level of service," base
traffic engineering on a GoS during the peak or busy hour. GoS is a
unit of measurement of the chance that a call is blocked. For example, a GoS of
P(.01) means that one call is blocked in 100 call attempts. A GoS of P(.001)
results in one blocked call per 1000 attempts. Look at call attempts during the
day's busiest hour. The most accurate method to find the busiest hour is to
take the ten busiest days in a year, sum the traffic on an hourly basis, find
the busiest hour, then derive the average amount of time.

In North America the 10 busiest days of the year are used to find
the busiest hour. Standards such as Q.80 and Q.87 use other methods to
calculate the busy hour. Use a number that is sufficiently large in order to
provide a GoS for busy conditions and not the average hour traffic.

The traffic volume in telephone engineering is measured in units
called erlangs. An erlang is the amount of traffic one
trunk handles in one hour. It is a non-dimensional unit that has many
functions. The easiest way to explain erlangs is through the use of an example.

Assume that you have eighteen trunks that carry nine erlangs of
traffic with an average duration of all calls of three minutes. What is the
average number of busy trunks, the number of call originations in one hour, and
the time it takes to complete all calls?

What is the average number of busy trunks?

With nine erlangs of traffic, nine trunks are busy since an
erlang is the amount of traffic one trunk handles in one hour.

What is the number of call originations in one hour?

Given that there are nine erlangs of traffic in one hour and an
average of three minutes per call, convert one hour to minutes, multiply the
number of erlangs, and divide the total by the average call duration. This
yields 180 calls.

Nine in one hour multiplied by 60 minutes/hour divided by
three minutes/call = 180 calls.

Erlangs are dimensionless. However, they are referenced to
hours.

What is the time it takes to complete all calls?

With 180 calls that last three minutes per call, the total time
is 540 minutes, or nine hours.

Other equivalent measurements that you can potentially encounter
include:

1 erlang =

60 call minutes =

3600 call seconds =

36 centum call seconds (CCS)

A simple way to calculate the busy hour is to collect one business
month's worth of traffic. Determine the amount of traffic that occurs in a day
based on twenty-two business days in a month. Multiply that number by fifteen
percent to seventeen percent. As a rule, the busy hour traffic represents
fifteen percent to seventeen percent of the total traffic that occurs in one
day.

Once you have determined the amount of traffic in erlangs that
occurs during the busy hour, the next step is to determine the number of trunks
required to meet a particular GoS. The number of trunks required differs based
on the traffic probability assumptions.

The first assumption is the number of potential sources. Sometimes,
there is a major difference between planning for an infinite versus a small
number of sources. For this example, ignore the method of how this is
calculated. The table here compares the amount of traffic the system needs to
carry in erlangs to the amount of potential sources offering traffic. It
assumes that the number of trunks holds constant at ten for a GoS of .01.

Only 4.13 erlangs are carried if there are an infinite number of
sources. The reason for this phenomenon is that as the number of sources
increases, the probability of a wider distribution in the arrival times and
holding times of calls increases. As the number of sources decreases, the
ability to carry traffic increases. At the extreme end, the system supports ten
erlangs. There are only ten sources. So, if sizing a PBX or key system in a
remote branch office, you can get by with fewer trunks and still offer the same
GoS.

Poisson Distribution with 10 trunks and a P of 0.01 *

Number of Sources

Traffic Capacity (erlangs)

Infinite

4.13

100

4.26

75

4.35

50

4.51

25

4.84

20

5.08

15

5.64

13

6.03

11

6.95

10

10

Note: The equations traditionally used in telephone engineering are based
on the Poisson arrival pattern. This is an approximate exponential
distribution. This exponential distribution indicates that a small number of
calls are very short in length, a large number of calls are only one to two
minutes in length. As the calls lengthen they decrease exponentially in number
with a very small number of calls over ten minutes. Although this curve does
not exactly duplicate an exponential curve, it is found to be quite close in
actual practice.

The second assumption deals with the traffic arrival characteristics.
Usually, these assumptions are based on a Poisson traffic distribution where
call arrivals follow a classic bell-shaped curve. Poisson distribution is
commonly used for infinite traffic sources. In the three graphs here, the
vertical axis shows the probability distribution and the horizontal axis shows
the calls.

Random Traffic

Bunched calls result in traffic that has a smooth-shaped pattern. This
pattern occurs more frequently with finite sources.

Smooth Traffic

Peaked or rough traffic is represented by a skewed shape. This
phenomenon occurs when traffic rolls from one trunk group to another.

How to handle lost calls is the third assumption. The figure here
depicts the three options available when the station you call does not answer:

Lost Calls Cleared (LCC).

Lost Calls Held (LCH).

Lost Calls Delayed (LCD).

The LCC option assumes that once a call is placed and the server
(network) is busy or not available, the call disappears from the system. In
essence, you stop and do something different.

The LCH option assumes that a call is in the system for the duration of
the hold time, regardless of whether or not the call is placed. In essence, you
continue to redial for as long as the hold time before you stop.

Recalling, or redialing, is an important traffic consideration. Assume
that 200 calls are attempted. Forty receive busy signals and attempt to redial.
That results in 240 call attempts, a 20% increase. The trunk group now provides
an even poorer GoS than initially thought.

The LCD option means that once a call is placed, it remains in a queue
until a server is ready to handle it. Then it uses the server for the full
holding time. This assumption is most commonly used for automatic call
distribution (ACD) systems.

The assumption that the lost calls clear the system tends to understate
the number of trunks required. On the other hand, LCH overstates the
number.

The fourth and final assumption centers around the switching equipment
itself. In the circuit switch environment, many of the larger switches block
switches. That is, not every input has a path to every output. Complex grading
structures are created to help determine the pathways a circuit takes through
the switch, and the impact on the GoS. In this example, assume that the
equipment involved is fully non-blocking.

The purpose of the third step is to calculate the number of physical
trunks required. You have determined the amount of offered traffic during the
busy hour. You have talked to the customer. Therefore, you know the GoS the
customer requests . ` Calculate the number of trunks required by using formulas
or tables.

Traffic theory consists of many queuing methods and associated
formulas. Tables that deal with the most commonly encountered model is
presented here. The most commonly used model and table is Erlang B. It is based
on infinite sources, LCC, and Poisson distribution that is appropriate for
either exponential or constant holding times. Erlang B understates the number
of trunks because of the LCC assumption. However, it is the most commonly used
algorithm.

The example here determines the number of trunks in a trunk group that
carry this traffic (a trunk group is defined as a hunt group of parallel
trunks):

Based on these assumptions, the appropriate algorithm to use is Erlang
B. Use this table to determine the appropriate number of trunks (N) for a P of
.01.

N

P

.003

.005

.01

.02

.03

.05

1

.003

.005

.011

.021

.031

.053

2

.081

.106

.153

.224

.282

.382

3

.289

.349

.456

.603

.716

.9

4

.602

.702

.87

1.093

1.259

1.525

5

.995

1.132

1.361

1.658

1.876

2.219

6

1.447

1.622

1.909

2.276

2.543

2.961

7

1.947

2.158

2.501

2.936

3.25

3.738

8

2.484

2.73

3.128

3.627

3.987

4.543

9

3.053

3.333

3.783

4.345

4.748

5.371

10

3.648

3.961

4.462

5.084

5.53

6.216

11

4.267

4.611

5.16

5.842

6.328

7.077

12

4.904

5.279

5.876

6.615

7.141

7.95

13

5.559

5.964

6.608

7.402

7.967

8.835

14

6.229

6.664

7.352

8.201

8.804

9.73

15

6.913

7.376

8.108

9.01

9.65

10.63

Note: Table is extracted from T. Frankel's "ABC of the Telephone"

Since a grade of service of P .01 is required, use only the column
designated as P .01. The calculations indicate a busy hour traffic amount of
2.64 erlangs. This lies between 2.501 and 3.128 in the P .01 column. This
corresponds to a number of trunks (N) of seven and eight. Since you are unable
to use a fractional trunk, use the next larger value ( eight trunks) to carry
the traffic.

There are several variations of Erlang B tables available to determine
the number of trunks required to service a specific amount of traffic. The
table here shows the relationship between GoS and the number of trunks (T)
required to support a rate of traffic in erlangs.

In most situations, a single circuit between units is enough for the
expected number of voice calls. However, in some routes there is a
concentration of calls that requires additional circuits to be added to provide
a better GoS. A GoS in telephone engineering usually ranges from 0.01 to 0.001.
This represents the probability of the number of calls that are blocked. In
other words, .01 is one call in 100, and .001 is one call in 1000 that is lost
due to blocking. The usual way to describe the GoS or blocking characteristics
of a system is to state the probability that a call is lost when there is a
given traffic load. P(01) is considered a good GoS, whereas P(001) is
considered a non-blocking GoS.

4. Determine the proper mix of trunks.

The proper mix of trunks is more of an economic decision than a
technical decision. Cost per minute is the most commonly used measurement in
order to determine the price breakpoint of adding trunks. Ensure that all cost
components are considered, such as accounting for additional transmission,
equipment, administration, and maintenance costs.

There are two rules to follow when you optimize the network for cost:

Use average usage figures instead of the busy hour which overstates
the number of call minutes.

Use the least costly circuit until the incremental cost becomes more
expensive than the next best route.

Based on the previous example, providing a
GoS of .01 requires 8 trunks if there are 2.64 erlangs of offered traffic.
Derive an average usage figure:

352 hours divided by 22 days in a month divided by 8 hours in a day x
1.10 (call processing overhead) = 2.2 erlangs during the average hour.

Assume that the carrier (XYZ) offers these rates:

Direct distance dialing (DDD) = $25 per hour.

Savings Plan A = $60 fixed charge plus $18 per hour.

Tie trunk = $500 flat rate.

First, graph the costs. All the numbers are converted to hourly figures
to make it easier to work with the erlang calculations.

The Tie Trunk, represented by the red line, is a straight line at
$500. DDD is a linear line that starts at 0. To optimize costs, the goal is to
stay below the curve. The cross-over points between the different plans occur
at 8.57 hours between DDD and Plan A, and 24.4 hours between Plan A and Tie
Trunks.

The next step is to calculate the carried traffic on a per trunk basis.
Most switches allocate voice traffic on a first-in-first-out (FIFO) basis. This
means that the first trunk in a trunk group carries substantially more traffic
than the last trunk in the same trunk group. Calculate the average allocation
of traffic per trunk. It is difficult to do so without a program that
calculates these figures on an iterative basis. This table shows the traffic
distribution based on 2.2 erlangs using such a program:

Traffic on Each Trunk Based on 2.2 Erlangs

Trunks

Offered Hours

Carried per Trunk

Cumulative Carried

GoS

1

2.2

0.688

0.688

0.688

2

1.513

0.565

1.253

0.431

3

0.947

0.419

1.672

0.24

4

0.528

0.271

1.943

0.117

5

0.257

0.149

2.093

0.049

6

0.107

0.069

2.161

0.018

7

0.039

0.027

2.188

0.005

8

0.012

0.009

2.197

0.002

9

0.003

0.003

2.199

0

The first trunk is offered 2.2 hours and carries .688 erlangs. The
theoretical maximum for this trunk is one erlang. The eighth trunk only carries
.009 erlangs. An obvious implication when you design a data network to carry
voice is that the specific trunk moved on to the data network can have a
considerable amount of traffic carried, or next to nothing carried.

Using these figures and combining them with the break even prices
calculated earlier, you can determine the appropriate mix of trunks. A trunk
can carry 176 erlangs of traffic per month, based on 8 hours per day and 22
days per month. The first trunk carries .688 erlangs or is 68.8% effective. On
a monthly basis, that equals 121 erlangs. The cross-over points are 24.4 and
8.57 hours. In this figure, tie trunks are still used at 26.2 erlangs. However,
the next lower trunk uses Plan A because it drops below 24.4 hours. The same
method applies to the DDD calculations.

Regarding voice over data networks, it is important to derive a cost
per hour for the data infrastructure. Then, calculate the voice over X trunk as
another tariffed option.

5. Equate erlangs of carried traffic to packets or cells per second.

The fifth and last step in traffic engineering is to equate erlangs of
carried traffic to packets or cells per second. One way to do this is to
convert one erlang to the appropriate data measurement, then apply modifiers.
These equations are theoretical numbers based on pulse code modulation (PCM)
voice and fully loaded packets.

Apply modifiers to these figures based on the actual conditions. Types
of modifiers to apply include packet overhead, voice compression, voice
activity detection (VAD), and signaling overhead.

Packet overhead can be used as a percent modifier.

ATM

AAL1 has nine bytes for every 44 bytes of payload or has a 1.2
multiplier.

AAL5 has six bytes for every 47 bytes of payload or has a 1.127
multiplier.

Frame Relay

Four to six bytes of overhead, payload variable to 4096
bytes.

Using 30 bytes of payload and four bytes of overhead, it has a 1.13
multiplier.

IP

20 bytes for IP.

Eight bytes for User Datagram Protocol (UDP).

Twelve to 72 bytes for Real-Time Transport Protocol
(RTP).

Without using Compressed Real-Time Protocol (CRTP), the amount of
overhead is unrealistic. The actual multiplier is three. CRTP can reduce the
overhead further, generally in the range of four to six bytes. Assuming five
bytes, the multiplier changes to 1.25. Assume that you run 8 KB of compressed
voice. You are unable to get below 10 KB if you factor in overhead. Consider
Layer 2 overhead as well.

Voice compression and voice activity detection are also treated as
multipliers. For example, conjugate structure algebraic code excited linear
prediction (CS-ACELP) ( 8 KB voice) is considered a .125 multiplier. VAD can be
considered a .6 or .7 multiplier.

Factor in signaling overhead. In particular, VoIP needs to figure in
the Real Time Control Protocol (RTCP) and the H.225 and H.245 connections.

The final step is to apply traffic distribution to the trunks to see
how it equates to bandwidth. This diagram shows the traffic distribution based
on busy hour and average hour calculations. For the busy hour calculations, the
program that shows the distribution of traffic per trunk based on 2.64 erlangs
is used.

BH = Busy Hour

AH = Average Hour

Using the average hour figures as an example, there are .688 erlangs on
the first trunk. This equates to 64 kBps x .688 = 44 kBps. 8 KB voice
compression equates to 5.5 kBps. IP overhead factored in brings the number up
to 6.875 kBps. With voice trunks, the initial trunks carry high traffic only in
larger trunk groups.

When you work with voice and data managers, the best approach to take
when you calculate voice bandwidth requirements is to work through the math.
Eight trunks are needed at all times for peak traffic intensity. Using PCM
voice results in 512 KB for eight trunks. The busy hour uses 2.64 erlangs, or
169 kBps of traffic. On average, you use 2.2 erlangs or 141 kBps of traffic.

2.2 erlangs of traffic carried over IP using voice compression requires
this bandwidth:

In today's customer private networks, attention must be given to
transmission parameters, such as end-to-end loss and propagation delay.
Individually, these characteristics hinder the efficient transfer of
information through a network. Together, they manifest themselves as an even
more detrimental obstruction referred to as "echo."

Loss is introduced into transmission paths between end offices (EO)
primarily to control echo and near-singing (Listener Echo). The amount of loss
needed to achieve a given talker-echo GoS increases with delay. However, the
loss also attenuates the primary speech signal. Too much loss makes it
difficult to hear the speaker. The degree of difficulty depends upon the amount
of noise in the circuit. The joint effect of loss, noise, and talker-echo is
assessed through the loss-noise-echo GoS measure. The development of a loss
plan takes into account the joint customer perception effect of the three
parameters (loss, noise, and talker echo). A loss plan needs to provide a value
of connection loss that is close to the optimum value for all connection
lengths. At the same time, the plan must be easy enough to implement and
administer. The information here helps you to design and implement the Cisco
MC3810 into a customer private network.

A PBX is an assembly of equipment that allows an individual within a
community of users to originate and answer calls to and from the public network
( through central office, wide-area telephone service (WATS), and FX trunks),
special service trunks, and other users (PBX lines) within the community. Upon
dial initiation, the PBX connects the user to an idle line or to an idle trunk
in an appropriate trunk group. It returns the appropriate call status signal,
such as a dial tone or audible ring. A busy indication is returned if the line
or trunk group is busy. An attendant position can be provided to answer
incoming calls and for user assistance. There are both Analog and Digital PBXs.
An Analog PBX (APBX) is a dial PBX that uses analog switching to make call
connections. A Digital PBX (DPBX) is a dial PBX that uses digital switching to
make call connections. PBXs function in one of three ways: Satellite, Main, and
Tandem.

A Satellite PBX is homed on a Main PBX through which it receives calls
from the public network and can connect to other PBXs in a private network.

A Main PBX functions as the interface to the Public Switched Telephone
Network (PSTN). It supports a specific geographic area. It can support a
subtending Satellite PBX as well as function as a Tandem PBX.

A Tandem PBX functions as a through-point. Calls from one Main PBX are
routed through another PBX to a third PBX. Therefore, the word Tandem.

The interfaces and levels expected by DPBXs are listed first in order
to help design and implement the Cisco MC3810s with the correct transmit and
receive levels. DPBXs with pure digital tie trunks (no analog-to-digital
conversions) always receive and transmit at 0 dB (D/TT), as illustrated in the
previous figure.

For DPBXs with hybrid tie trunks (analog-to-digital conversion), the
transmit and receive levels are also 0 dB if the Channel Bank (CB) interface
connects to the DPBX digitally at both ends and an Analog Tie Trunk is used
(see the next figure). If the CB connects to the DPBX through an analog
interface, the levels are -2.0 dB for both transmit and receive (see this
figure).

DPBXs with Hybrid Tie Trunks

Channel Bank Connects to the DPBX Through an Analog
Interface

If there is only one CB and it connects to a DPBX through an analog
interface, the levels are -2.0 dB transmit and -4.0 receive (see this figure).

When you implement Cisco MC3810s into a customer network, you must
first understand the existing network loss plan to ensure that an end-to-end
call still has the same overall loss or levels when the Cisco MC3810s are
installed. This process is called baselining or benchmarking. One way to
benchmark is to draw all of the network components before you install the Cisco
MC3810. Then document the expected levels at key access and egress points in
the network, based on Electronic Industries Association and Telecommunications
Industry Association (EIA/TIA) standards. Measure the levels at these same
access and egress points in the network to ensure that they are properly
documented (see this figure). Once the levels are measured and documented,
install the Cisco MC3810. Once installed, adjust the levels of the Cisco MC3810
to match the levels previously measured and documented (see this figure).

Network Components Before you Install the Cisco MC3810

Network Components After you Install the Cisco MC3810

For the majority of Cisco MC3810 implementations, DPBXs are part of the
overall customer network. For example, the network topology can look like this:

DPBX (Location 1) connects to a Cisco MC3810 (Location 1). This
connects to a facility/trunk (digital or analog) to a distant end (Location 2).
The facility/trunk is connected to another Cisco MC3810. This is connected to
another DPBX (Location 2). In this scenario, the levels (transmit and receive)
that are expected at the DPBX are determined by the facility/trunk type or
interface (as illustrated in the previous figure).

The next step is to start the design:

Diagram the existing network with all of the transmission equipment
and facility connections included.

Measure the actual levels to ensure that the expected levels and the
actual levels are the same. If they are not, go back and review the EIA/TIA
documents for the type of configuration and interface. Make level adjustments
as necessary. If they are the same, document the levels and move on to the next
piece of equipment. Once you have documented all of the measured levels in the
network and they are consistent with the expected levels, you are ready to
install the Cisco MC3810.

Install the Cisco MC3810 and adjust the levels to match the levels
measured and documented prior to installation. This ensures that the overall
levels are still consistent with those of the benchmark levels. Make a call
through test to ensure the Cisco MC3810 operates efficiently. If not, go back
and recheck the levels to ensure they are set correctly.

The Cisco MC3810 can also be used to interface to the PSTN. It is
designed to have - 3 dB on Foreign Exchange Station (FXS) ports, and 0 dB for
Foreign Exchange Office (FXO) and recEive and transMit (E&M) ports. For
analog, these values are true for both directions. For digital, the value is 0
dB. The Cisco MC3810 has a dynamic command to show the actual gain
(show voice call x/y) to allow a technician to hold
a digit key and watch the actual gain for various DTMF tones.

The hierarchical synchronization method consists of four stratum levels
of clocks. It is selected to synchronize the North American networks. It is
consistent with the current industry standards.

In the hierarchical synchronization method, frequency references are
transmitted between nodes. The highest level clock in the synchronization
hierarchy is a Primary Reference Source (PRS). All interconnecting digital
synchronization networks need to be controlled by a PRS. A PRS is equipment
that maintains a long-term frequency accuracy of 1x10-11 or better with
optional verification to Coordinated Universal Time (UTC) and meets current
industry standards. This equipment can be a stratum 1 clock (Cesium standard)
or can be equipment directly controlled by standard UTC-derived frequency and
time services, such as LORAN-C or Global Positioning Satellite System (GPS)
radio receivers. The LORAN-C and GPS signals themselves are controlled by
Cesium standards that are not a part of the PRS since they are physically
removed from it. Because primary reference sources are stratum 1 devices or are
traceable to stratum 1 devices, every digital synchronization network
controlled by a PRS has stratum 1 traceability.

One attractive feature of hierarchical synchronization is that existing
digital transmission facilities between digital switching nodes can be used for
synchronization. For example, the basic 1.544 MB/s line rate
(8000-frame-per-second frame rate) of a T1 Carrier System can be used for this
purpose without diminishing the traffic carrying capacity of that carrier
system. Hence, separate transmission facilities do not need to be dedicated for
synchronization. However, synchronization interfaces between public and private
networks need to be coordinated because of certain digital transmission
facility characteristics, such as facility trouble history, pointer
adjustments, and the number of switching points.

Reliable operation is crucial to all parts of a telecommunications
network. For this reason, the synchronization network includes primary and
secondary (backup) synchronization facilities to each Stratum 2 node, many
Stratum 3 nodes, and Stratum 4 nodes, where applicable. In addition, each
Stratum 2 and 3 node is equipped with an internal clock that bridges short
disruptions of the synchronization references. This internal clock is normally
locked to the synchronization references. When the synchronization reference is
removed, the clock frequency is maintained at a rate determined by its
stability.

Private digital networks, when interconnected with PRS-traceable local
exchange carrier/ International Electrotechnical Commission (LEC/IEC) networks,
need to be synchronized from a reference signal traceable to a PRS. Two methods
can be employed to achieve PRS traceability:

Provide a PRS clock, in which case the network operates
plesiochronously with the LEC/IEC networks.

There are fundamentally two architectures that can be used to pass
timing across the interface between LEC/IEC and the private network. The first
is for the network to accept a PRS-traceable reference from an LEC/IEC at one
location and to then provide timing references to all other equipment over
interconnecting facilities. The second is for the network to accept a
PRS-traceable reference at each interface with an LEC/IEC.

In the first method, the private network has control of the
synchronization of all equipment. However, from a technical and maintenance
viewpoint, there are limitations. Any loss of the distribution network causes
all of the associated equipment to slip against the LEC/IEC networks. This
problem causes troubles that are difficult to detect.

In the second method, PRS-traceable references are provided to the
private network at each interface with an LEC/IEC. In this arrangement, the
loss of a PRS-traceable reference causes a minimum of troubles. Additionally,
the slips against the LEC/IEC occur at the same interface as the source of the
trouble. This makes trouble location and subsequent repairs easier.

Signaling is defined by CCITT Recommendation Q.9 as "the exchange of
information (other than speech) specifically concerned with the establishment,
release, and control of calls, and network management in automatic
telecommunications operations."

In the broadest sense, there are two signaling realms:

Subscriber signaling

Trunk signaling (interswitch and/or interoffice)

Signaling is also traditionally classified into four basic functions:

Supervision

Address

Call Progress

Network Management

Supervision signaling is used to:

Initiate a call request on line or trunks (called line signaling on
trunks)

Hold or release an established connection

Initiate or terminate charging

Recall an operator on an established connection

Address signaling conveys such information as the calling or called
subscriber's telephone number and an area code, an access code, or a Private
Automatic Branch Exchange (PABX) tie trunk access code. An address signal
contains information that indicates the destination of a call initiated by a
customer, network facility, and so forth.

Call progress signals are usually audible tones or recorded
announcements that convey call-progress or call-failure information to
subscribers or operators. These call-progress signals are fully described
.

Network management signals are used to control the bulk assignment of
circuits or to modify the operating characteristics of switching systems in a
network in response to overload conditions.

There are about 25 recognized interregister signaling systems
worldwide, in addition to some subscriber signaling techniques. CCITT Signaling
System Number 7 (SSN7) is fast becoming the international/national standard
interregister signaling system.

Most installations will probably involve E&M signaling. However,
for reference, single frequency (SF) signaling on Tip and Ring loops, Tip and
Ring reverse battery loops, loop start, and ground start are also included.

Types I and II are the most popular E&M signaling in the Americas.
Type V is used in the United States. It is also very popular in Europe. SSDC5A
differs in that on- and off-hook states are reversed to allow for fail-safe
operation. If the line breaks, the interface defaults to off-hook (busy). Of
all the types, only II and V are symmetrical ( can be back-to-back using a
cross-over cable). SSDC5 is most often found in England.

Other signaling techniques often used are delay, immediate, and wink
start. Wink start is an in-band technique where the originating device waits
for an indication from the called switch before it sends the dialed digits.
Wink start normally is not used on trunks that are controlled with
message-oriented signaling schemes such as ISDN or Signaling System 7 (SS7).

SF in-band signaling is widely used in North America. Its most common
application is for supervision, such as idle-busy, also called line signaling.
It can also be used for dial pulse signaling on trunks. The dynamics of SF
signaling requires an understanding of the signal durations and configurations
of the E&M circuits, as well as the lead interface arrangements. These
tables show the characteristics of SF signaling, E&M lead configurations,
and interface arrangements.

Typical Single Frequency Signaling
Characteristics

General

Signaling frequency (tone)

2600 Hz

Idle state transmission

Cut

Idle/break

Tone

Busy/make

No Tone

Receiver

Detector bandwidth

+/- 50 Hz @ -7 dBm for E type
+/- 30 Hz @ -7 dBm

Pulsing rate

7.5 to 122 pps

E/M unit

Minimum time for on-hook

33 ms

Minimum no tone for off-hook

55 ms

Input percent break (tone)

38-85 (10 pps)

E lead - open

Idle

- ground

Busy

Originating (loop reverse battery) unit

Minimum tone for idle

40 ms

Minimum no tone for off-hook

43 ms

Minimum output for on-hook

69 ms

Voltage on R lead (-48 V on ring and ground on
tip)

On-hook

Voltage on T lead (-48 V on tip and ground on
ring)

Off-hook

Terminating (loop reverse battery) unit

Minimum tone for on-hook

90 ms

Minimum no tone for off-hook

60 ms

Minimum output (tone-on)

56 ms

Loop open

On-hook

Loop closed

Off-hook

Transmitter

Low level tone

-36 dBm

High level tone

-24 dBm

High level tone duration

400 ms

Precut

8 ms

Holdover cut

125 ms

Crosscut

625 ms

On hook cut

625 ms

E/M unit

Voltage on M lead

Off-hook (no tone)

Open/ground on M lead

On-hook (tone)

Minimum ground on M lead

21 ms

Minimum voltage on M lead

21 ms

Minimum output tone

21 ms

Minimum no tone

21 ms

Originating (loop reverse battery) unit

Loop current to no tone

19 ms

No loop current to tone

19 ms

Minimum input for tone out

20 ms

Minimum input for no tone out

14 ms

Minimum tone out

51 ms

Minimum no tone out

26 ms

Loop open

On-hook

Loop closed

Off-hook

Terminating (loop) unit

Reverse battery to no tone

19 ms

Normal battery to tone

19 ms

Minimum battery for tone out

25 ms

Minimum reverse battery for no tone

14 ms

Minimum tone out

51 ms

Minimum no tone out

26 ms

Battery on R lead (-48 v)

On-hook

Battery on TY lead (-48 on tip

Off-hook

Single Frequency Signals Used in E&M Lead
Signaling

Calling End

Called End

Signal

M-Lead

E-Lead

2600 Hz

2600 Hz

E-Lead

M-Lead

Signal

Idle

Ground

Open

On

On

Open

Ground

Idle

Connect

Battery

Open

Off

On

Ground

Ground

Connect

Stop dialing

Battery

Ground

Off

Off

Ground

Battery

Stop dialing

Start dialing

Battery

Open

Off

On

Ground

Ground

Start Dialing

Dial pulsing

Ground

Open

On

On

Open

Ground

Dial pulsing

Battery

Off

Ground

Off -hook

Battery

Ground

Off

Off

Ground

Battery

Off-hook (answer)

Ring forward

Ground

Ground

On

Off

Open

Battery

Ring forward

Battery

Off

Ground

Ringback

Battery

Open

Off

On

Ground

Ground

Ringback

Ground

Off

Battery

Flashing

Battery

Open

Off

On

Ground

Ground

Flashing

Ground

Off

Battery

On-hook

Battery

Open

Off

On

Ground

Ground

On-hook

Disconnect

Ground

Open

On

On

Open

Ground

Disconnect

Single Frequency Signals Used in Reverse Battery Tip and Ring
Loop Signaling

Calling End

Called End

Signal

T/R - SF

SF - T/R

2600 Hz

2600 Hz

T/R - SF

SF - T/R

Signal

Idle

Open

Batt-gnd

On

On

Open

Batt-gnd

Idle

Connect

Closure

Batt-gnd

Off

On

Closure

Batt-gnd

Connect

Stop dialing

Closure

Rev batt-gnd

Off

Off

Closure

Rev batt-gnd

Stop dialing

Start dialing

Closure

Batt-gnd

Off

On

Closure

Batt-gnd

Start dialing

Dial pulsing

Open

Batt-gnd

On

On

Open

Batt-gnd

Dial pulsing

Closure

Off

Closure

Off-hook

Closure

Rev batt-gnd

Off

Off

Closure

Rev batt-gnd

Off-hook (answer)

Ring forward

Open

Rev batt-gnd

On

Off

Open

Rev batt-gnd

Ring forward

Closure

Off

Closure

Ringback

Closure

Batt-gnd

Off

On

Closure

Batt-gnd

Ringback

Rev batt-gnd

Off

Rev batt-gnd

Flashing

Closure

Batt-gnd

Off

On

Closure

Batt-gnd

Flashing

Rev batt-gnd

Off

Rev batt-gnd

On-hook

Closure

Batt-gnd

Off

On

Closure

Batt-gnd

On-hook

Disconnect

Open

Batt-gnd

On

On

Open

Batt-gnd

Disconnect

Single Frequency Signals Used for Ringing and Loop-Start
Signaling Using Tip and Ring Leads - Call Originating at Central Office
End

Signal

T/R - SF

SF - T/R

2600 Hz

2600 Hz

T/R - SF

SF - T/R

Signal

Idle

Gnd-batt

Open

Off

On

Gnd-batt

Open

Idle

Seizure

Gnd-batt

Open

Off

On

Gnd-batt

Open

Idle

Ringing

Gnd-batt and 20 Hz

Open

On-off

On

Gnd-batt and 20 Hz

Open

Ringing

Off-hook (ring-trip and talk)

Gnd-batt

Closure

Off

Off

Gnd-batt

Closure

Off-hook (ring-trip and answer)

On-hook

Gnd-batt

Closure

Off

Off

Gnd-batt

Closure

Off-hook

On-hook (hang-up)

Gnd-batt

Open

Off

On

Gnd-batt

Open

On-hook (hang-up)

Note: 20 Hz ringing (2 sec on, 4 sec off)

Single Frequency Signals Used for Ringing and Loop-Start
Signaling Using Tip and Ring Leads - Call Originating at Station End

Signal

T/R - SF

SF - T/R

2600 Hz

2600 Hz

T/R - SF

SF - T/R

Signal

Idle

Open

Gnd-batt

On

Off

Open

Gnd-batt

Idle

Off-hook (seizure)

Closure

Gnd-batt

Off

Off

Closure

Gnd-batt

Idle

Start dial

Closure

Dial tone and gnd-batt

Off

Off

Closure

Dial tone and gnd-batt

Start dial

Dial pulsing

Open-closure

Gnd-batt

On-off

Off

Open-closure

Gnd-batt

Dial pulsing

Waiting answer

Closure

Audible ring and gnd-batt

Off

Off

Closure

Audible ring and gnd-batt

Waiting answer

On-hook (talk)

Closure

Gnd-batt

Off

Off

Closure

Gnd-batt

Off-hook (answered)

On-hook (hang up)

Open

Gnd-batt Closure

On

Off

Open

Gnd-batt

On-hook (disconnected) Off-hook

Single Frequency Signals Used for Ringing and Ground-Start
Signaling Using Tip and Ring Leads - Call Originating at Central Office
End

Signal

T/R - SF

SF - T/R

2600 Hz

2600 Hz

T/R - SF

SF - T/R

Signal

Idle

Open-batt

Batt-batt

On

On

Open-batt

Idle

Seizure

Gnd-batt

Open

On

On

Gnd-batt

Make-busy

Ringing

Gnd-batt and 20 Hz

Open

On and 20 Hz

On

Gnd-batt and 20 Hz

Open

Ringing

Off-hook (ring-trip and talk)

Gnd-batt

Closure

Off

Off

Gnd-batt

Closure

Off-hook (ring-trip and answer)

On-hook

Gnd-batt

Closure

On

Off

Open-batt

Closure

On-hook

On-hook (hang-up)

Gnd-batt

Open

Off

On

Gnd-batt

Open

On-hook (hang-up)

Note: 20 Hz ringing (2 sec on, 4 sec off)

Single Frequency Signals Used for Ringing and Ground-Start
Signaling Using Tip and Ring Leads - Call Originating at Station
End

The Cisco MC3810 supports the concept of hunting groups. This is the
configuration of a group of dial peers on the same PBX with the same
destination pattern. With a hunting group, if a call attempt is made to a dial
peer on a specific digital signal level 0 (DS-0) timeslot and that timeslot is
busy, the Cisco MC3810 hunts for another timeslot on that channel until an
available timeslot is found. In this case, each dial peer is configured using
the same destination pattern of 3000. It forms a dial pool to that destination
pattern. To provide specific dial peers in the pool with a preference over
other dial peers, configure the preference order for each dial peer using the
preference command. The preference value is between
zero and ten. Zero means the highest priority. This is an example of the dial
peer configuration with all dial peers having the same destination pattern, but
with different preference orders:

You can also set the preference order on the network side for
voice-network dial peers. However, you cannot mix the preference orders for
POTS dial peers (local telephone devices) and voice-network peers (devices
across the WAN backbone). The system only resolves the preference among dial
peers of the same type. It does not resolve preferences between the two
separate preference order lists. If POTS and voice-network peers are mixed in
the same hunt group, the POTS dial peers must have priority over the
voice-network peers. To disable further dial peer hunting if a call fails, the
huntstop configuration command is used. To reenable
it, the nohuntstop command is used.

The acceptance plan needs to contain elements that demonstrate the
dial/numbering plan and all voice quality issues such as the gain/loss plan,
traffic engineering or loading, and signaling and interconnection with all
equipment.

Verify that the voice connection works by doing these :

Pick up the handset of a telephone connected to the
configuration. Verify that there is a dial tone.

Make a call from the local telephone to a configured dial peer.
Verify that the call attempt is successful.

Check the validity of the dial peer and voice port configuration by
performing these tasks:

If you have relatively few dial peers configured, use the
show dial-peer voice summary command to verify that
the data configured is correct.

To show the status of the voice ports, use the show
voice port command.

To show the call status for all voice ports, use the
show voice call command.

To show the current status of all domain specific part (DSP)
voice channels, use the show voice dsp command.

If you have trouble connecting a call, try to resolve the problem by
performing these tasks:

If you suspect the problem is in the Frame Relay configuration, make
sure that frame-relay traffic-shaping is turned on.

If you send voice over Frame Relay traffic over serial port 2 with a
T1 controller, make sure the channel group command
is configured.

If you suspect the problem is associated with the dial peer
configuration, use the show dial-peer voice command
on the local and remote concentrators to verify that the data is configured
correctly on both.